[asterisk-bugs] [JIRA] (ASTERISK-24624) Transfer to invalid extension results in hung channel.

[Mark Michelson (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24624?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224432#comment-224432 ] 

Mark Michelson commented on ASTERISK-24624:
-------------------------------------------

I tried to reproduce this and it was not working for me. I was using revision 430587 of the Asterisk 13 branch. My CLI output looks almost identical to yours except that I don't have a hung channel:

{noformat}
*CLI>     -- Executing [201 at default:1] NoOp("PJSIP/202-00000003", "") in new stack
    -- Executing [201 at default:2] Dial("PJSIP/202-00000003", "PJSIP/201,,tT") in new stack
    -- Called PJSIP/201
    -- PJSIP/201-00000004 is ringing
    -- PJSIP/201-00000004 answered PJSIP/202-00000003
    -- Channel PJSIP/201-00000004 joined 'simple_bridge' basic-bridge <4141887b-0eb9-41ad-9f77-7ad8b80dbf83>
    -- Channel PJSIP/202-00000003 joined 'simple_bridge' basic-bridge <4141887b-0eb9-41ad-9f77-7ad8b80dbf83>
    -- Started music on hold, class 'default', on channel 'PJSIP/202-00000003'
    -- Stopped music on hold on PJSIP/202-00000003
    -- Channel PJSIP/201-00000004 left 'simple_bridge' basic-bridge <4141887b-0eb9-41ad-9f77-7ad8b80dbf83>
    -- Channel PJSIP/202-00000003 left 'simple_bridge' basic-bridge <4141887b-0eb9-41ad-9f77-7ad8b80dbf83>
    -- Executing [310 at default:1] VoiceMail("PJSIP/202-00000003", "NONEXISTENT_MAILBOX") in new stack
[2015-01-13 18:22:55.085] WARNING[18343][C-00000002]: app_voicemail.c:6467 leave_voicemail: No entry in voicemail config file for 'NONEXISTENT_MAILBOX'
    -- Auto fallthrough, channel 'PJSIP/202-00000003' status is 'ANSWER'

*CLI> 
*CLI> 
*CLI> core show channels
Channel              Location             State   Application(Data)             
0 active channels
0 active calls
3 calls processed
{noformat}

Out of curiosity, how are you performing your blind transfer? Are you using the transfer button on your SIP phone or are you pressing a DTMF sequence to do it? Does the hung channel happen every time or only sometimes?

> Transfer to invalid extension results in hung channel.
> ------------------------------------------------------
>
>                 Key: ASTERISK-24624
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24624
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.0.0, 13.0.1, 13.0.2, 13.1.0
>            Reporter: Zane Conkle
>            Assignee: Mark Michelson
>
> When attempting to blind transfer a call to an invalid extension or resource the channel will hang in the "Ring" state.
> I was able to duplicate this by blind transferring a call to a mailbox that does not exist. Below is the CLI output along with the stuck channel.
> {code}    -- Executing [s200-DEFAULT at default:1] Dial("SIP/carrier1-00000002", "PJSIP/200_default,10")
>     -- Called PJSIP/200_default
>     -- PJSIP/200_default-00000002 is ringing
>     -- PJSIP/200_default-00000002 answered SIP/carrier1-00000002
>     -- Channel SIP/carrier1-00000002 joined 'simple_bridge' basic-bridge <44f9110a-2508-4e62-acb4-75beca76ce31>
>     -- Channel PJSIP/200_default-00000002 joined 'simple_bridge' basic-bridge <44f9110a-2508-4e62-acb4-75beca76ce31>
>     -- Started music on hold, class 'default', on channel 'SIP/carrier1-00000002'
>     -- Stopped music on hold on SIP/carrier1-00000002
>     -- Channel SIP/carrier1-00000002 left 'simple_bridge' basic-bridge <44f9110a-2508-4e62-acb4-75beca76ce31>
>     -- Channel PJSIP/200_default-00000002 left 'simple_bridge' basic-bridge <44f9110a-2508-4e62-acb4-75beca76ce31>
>     -- Executing [1234 at default:1] VoiceMail("SIP/carrier1-00000002", "5678 at default") in new stack
> [Dec 16 10:29:58] WARNING[3204][C-00000002]: app_voicemail.c:6467 leave_voicemail: No entry in voicemail config file for '5678'
>     -- Auto fallthrough, channel 'SIP/carrier1-00000002' status is 'ANSWER'
> [root at pbx ~]# asterisk -rx "core show channels verbose" 
> Channel              Context              Extension        Prio State   Application  Data                      CallerID        Duration Accountcode PeerAccount BridgeID            
> PJSIP/200_default    default              s                   1 Ring    (None)       (Empty)                   200             00:27:57 default 
> {code}



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